• Journal of information and communication convergence engineering
• /
• v.10 no.2
• /
• pp.175-180
• /
• 2012

This paper presents a blind watermark detection scheme for an additive watermark embedding model. The proposed estimation-correlation-based watermark detector first estimates the embedded watermark by exploiting non-Gaussian of the real-world audio signal and the mutual independence between the host-signal and the embedded watermark and then a correlation-based detector is used to determine the presence or the absence of the watermark. For watermark estimation, blind source separation (BSS) based on independent component analysis (ICA) is used. Low watermark-to-signal ratio (WSR) is one of the limitations of blind detection with the additive embedding model. The proposed detector uses two-stage processing to improve the WSR at the blind detector; the first stage removes the audio spectrum from the watermarked audio signal using linear predictive (LP) filtering and the second stage uses the resulting residue from the LP filtering stage to estimate the embedded watermark using BSS based on ICA. Simulation results show that the proposed detector performs significantly better than existing estimation-correlationbased detection schemes.

• Proceedings of the IEEK Conference
• /
• 2003.11b
• /
• pp.243-246
• /
• 2003

In this paper, we propose a new digital audio watermarking technique with the wavelet transform. The watermark is embedded by eliminating unnecessary information of audio signal based on human auditory system (HAS). This algorithm is an audio watermarking method, which does not require any original audio information in watermark extraction process. In this paper, the masking effect is used for audio watermarking, that is, post-tempera] masking effect. We construct the window with the synchronization signal and we extract the best frame in the window by using the zero-crossing rate (ZCR) and the energy of the audio signal. The watermark may be extracted by using the correlation of the watermark signal and the portion of the frame. Experimental results show good robustness against MPEG1-layer3 compression and other common signal processing manipulations. All the attacks are made after the D/A/D conversion.

• Journal of information and communication convergence engineering
• /
• v.5 no.3
• /
• pp.245-248
• /
• 2007

In this paper, we research the content-based analysis and classification according to the composition of the feature parameter data base for the audio data to implement the audio data index and searching system. Audio data is classified to the primitive various auditory types. We described the analysis and feature extraction method for the feature parameters available to the audio data classification. And we compose the feature parameters data base in the index group unit, then compare and analyze the audio data centering the including level around and index criterion into the audio categories. Based on this result, we compose feature vectors of audio data according to the classification categories, and simulate to classify using discrimination function.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.17 no.2
• /
• pp.225-232
• /
• 2017

Recently, the development of digital transmission technology of audio signals and the introduction of audio network equipment using digital transmission technology have been made. As a result, audio network technology and equipment are actively applied to the design and construction of audio systems. The meeting room is a place where a large number of participants exchange opinions and communicate with each other. In addition to using an electric acoustic device such as a microphone and a speaker, it improves the intelligibility of the conference room through an example using an audio network.

• The Journal of the Acoustical Society of Korea
• /
• v.22 no.8
• /
• pp.687-693
• /
• 2003

In this paper, a new adaptive audio signal enhancement algorithm is proposed. In order to remove a broadband noise from a noisy signal, a filter is designed and applied adaptively to noisy audio signal. The noisy signal is first transformed to frequency domain and divided into bark domain to calculate excitation energy. A filter will be calculated to eliminate the noise by using the excitation energy and noisy energy which is obtained from a silent area. The filter is adaptively adjusted and continuously applied until the threshold point is met. The algorithm also works well even though the noise's energy change all of a sudden. SNR, NMR comparison and MOS Test are performed to show the effectiveness of the proposed algorithm.

• Journal of the Institute of Convergence Signal Processing
• /
• v.15 no.1
• /
• pp.1-8
• /
• 2014

Digital watermarking techniques are attracting attention as a proper solution to protect copyright for multimedia data. This paper proposes a new audio watermarking method based on Discrete Cosine Transformation (DCT) and Discrete Wavelet Transformation (DWT) for copyright protection. In our proposed watermarking method, the original audio is transformed into DCT domain and divided into two parts. Synchronization code is applied on the signal in first part and 2 levels DWT domain is applied on the signal in second part. The absolute value of DWT coefficient is divided into arbitrary number of segments and calculates the energy of each segment and middle peak. Watermarks are then embedded into each middle peak. Watermarks are extracted by performing the inverse operation of watermark embedding process. Experimental results show that the hidden watermark data is robust to re-sampling, low-pass filtering, re-quantization, MP3 compression, cropping, echo addition, delay, and pitch shifting, amplitude change. Performance analysis of the proposed scheme shows low error probability rates.

• Journal of Information Processing Systems
• /
• v.12 no.3
• /
• pp.525-537
• /
• 2016

Data hiding is a wide field that is helpful to secure network communications. It is common that many data hiding researchers consider improving and increasing many aspects such as capacity, stego file quality, or robustness. In this paper, we use an audio file as a cover and propose a reversible steganographic method that is modifying the sample values using modulus function in order to make the reminder of that particular value to be same as the secret bit that is needed to be embedded. In addition, we use a location map that locates these modified sample values. This is because in reversible data hiding it needs to exactly recover both the secret message and the original audio file from that stego file. The experimental results show that, this method (measured by correlation algorithm) is able to retrieve exactly the same secret message and audio file. Moreover, it has made a significant improvement in terms of the following: the capacity since each sample value is carrying a secret bit. The quality measured by peak signal-to-noise ratio (PSNR), signal-to-noise ratio (SNR), Pearson correlation coefficient (PCC), and Similarity Index Modulation (SIM). All of them have proven that the quality of the stego audio is relatively high.

• Proceedings of the KIEE Conference
• /
• 2004.11c
• /
• pp.605-607
• /
• 2004

A class D digital audio amplifier with small size, low cost, and high quality is positively necessary in the multimedia era made of home theater system and the digital audio broadcasting (DAB). It is impossible to analyze the stability of the digital audio amplifier, which is based on the PWM signal processing. To solve this problem, the digital audio amplifier is analyzed using variable structure control theory which is one of nonlinear system theories. Moreover, the magnitude and the frequency of ripple signal, which generated by hysteresis in the comparator, is obtained using describing function which is useful to represent the input-output relation of nonlinear system.

• Journal of the Korea Convergence Society
• /
• v.7 no.3
• /
• pp.77-85
• /
• 2016

We propose a digital signal processing technic that can compensate the non-linearity inherent in audio amplifiers, and present the result of the simulation. The inherent non-linearity of the audio power amplifier arising from analog devices is compensated via a digital signal processing technic consisting of indirect learning architecture and an adaptive filter. The simulation results show that the compensator can be realized using a third-order polynomial and compensates odd-order non-linearity efficiently. The even-oder non-linearity is mainly due to the dc offset at the output, which is difficult to eliminate with the proposed method. Care must be taken in designing the bias circuit to avoid the DC offset at the output. The proposed technic has significance in that digital signal processing technic can compensate for the impairment that is an inherent characteristic of an analog system.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.8 no.5
• /
• pp.35-41
• /
• 2008

This paper presents the design method of current-mode signal processing for high speed and low power digital audio signal processing. The digital audio processor requires a digital signal processing such as fast Fourier transform (FFT), which has a problem of large power consumption according to the settled point number and high speed operation. Therefore, a current-mode signal processing with a switched Current (SI) circuit was employed to the digital audio signal processing because a limited battery life should be considered for a low power operation. However, current memory that construct a SI circuit has a problem called clock-feedthrough. In this paper, we examine the connection of dummy MOS that is the common solution of clock-feedthrough and are willing to calculate the relation of width between dummy MOS for a proposal of the design methodology for improvement of current memory. As a result of simulation, in case of that the width of memory MOS is 20um, ratio of input current and bias current is 0.3, the relation of width between switch MOS and dummy MOS is $W_{M4}=1.95W_{M3}+1.2$ for the width of switch MOS is 2~5um, it is $W_{M4}=0.92W_{M3}+6.3$ for the width of switch MOS is 5~10um. Then the defined relation of MOS transistors can be a useful design guidance for a high speed low power digital audio processor.